similar to: h323 - SIP conversion

I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the

Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could

Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: ***************************************

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it ???? I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone:

I need some help with getting the following to work SipPhone <------> Asterisk <------> H323 GK (quintum) And H323Phone <------> Asterisk <------> H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk and Asterisk to the H323 GK. But when I try to make a call from

Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to myself I set tcpdump on the * box to capture all UDP traffic going to and from the ATA-186. If I

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP -

Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone

hello everybody i want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in 145: [peer146] host=192.168.0.146 type=friend context=from-trunk [to-146] type=peer host=192.168.0.146 faststart=yes tunneling=no progress_audio=yes disallow=all allow=alaw allow=ulaw this is mu extensions.conf

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm

Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this?

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you,

Hi All, I connect SIP phones and H323 phones, with *. But, if I call from H323 to SIP phone, i can hear nothing. I'm using h323 library. I found some e-mails at the Digium's Mail-list with the same problem. But, I coudn?t find the solution yet. Could you please help me?, any idea?. It's quite urgent. Thanks. -------------- next part -------------- An HTML attachment was